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  1. /*
  2.  * Opus encoder using libopus
  3.  * Copyright (c) 2012 Nathan Caldwell
  4.  *
  5.  * This file is part of FFmpeg.
  6.  *
  7.  * FFmpeg is free software; you can redistribute it and/or
  8.  * modify it under the terms of the GNU Lesser General Public
  9.  * License as published by the Free Software Foundation; either
  10.  * version 2.1 of the License, or (at your option) any later version.
  11.  *
  12.  * FFmpeg is distributed in the hope that it will be useful,
  13.  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  14.  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  15.  * Lesser General Public License for more details.
  16.  *
  17.  * You should have received a copy of the GNU Lesser General Public
  18.  * License along with FFmpeg; if not, write to the Free Software
  19.  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  20.  */
  21.  
  22. #include <opus.h>
  23. #include <opus_multistream.h>
  24.  
  25. #include "libavutil/opt.h"
  26. #include "avcodec.h"
  27. #include "bytestream.h"
  28. #include "internal.h"
  29. #include "libopus.h"
  30. #include "vorbis.h"
  31. #include "audio_frame_queue.h"
  32.  
  33. typedef struct LibopusEncOpts {
  34.     int vbr;
  35.     int application;
  36.     int packet_loss;
  37.     int complexity;
  38.     float frame_duration;
  39.     int packet_size;
  40.     int max_bandwidth;
  41. } LibopusEncOpts;
  42.  
  43. typedef struct LibopusEncContext {
  44.     AVClass *class;
  45.     OpusMSEncoder *enc;
  46.     int stream_count;
  47.     uint8_t *samples;
  48.     LibopusEncOpts opts;
  49.     AudioFrameQueue afq;
  50. } LibopusEncContext;
  51.  
  52. static const uint8_t opus_coupled_streams[8] = {
  53.     0, 1, 1, 2, 2, 2, 2, 3
  54. };
  55.  
  56. /* Opus internal to Vorbis channel order mapping written in the header */
  57. static const uint8_t opus_vorbis_channel_map[8][8] = {
  58.     { 0 },
  59.     { 0, 1 },
  60.     { 0, 2, 1 },
  61.     { 0, 1, 2, 3 },
  62.     { 0, 4, 1, 2, 3 },
  63.     { 0, 4, 1, 2, 3, 5 },
  64.     { 0, 4, 1, 2, 3, 5, 6 },
  65.     { 0, 6, 1, 2, 3, 4, 5, 7 },
  66. };
  67.  
  68. /* libavcodec to libopus channel order mapping, passed to libopus */
  69. static const uint8_t libavcodec_libopus_channel_map[8][8] = {
  70.     { 0 },
  71.     { 0, 1 },
  72.     { 0, 1, 2 },
  73.     { 0, 1, 2, 3 },
  74.     { 0, 1, 3, 4, 2 },
  75.     { 0, 1, 4, 5, 2, 3 },
  76.     { 0, 1, 5, 6, 2, 4, 3 },
  77.     { 0, 1, 6, 7, 4, 5, 2, 3 },
  78. };
  79.  
  80. static void libopus_write_header(AVCodecContext *avctx, int stream_count,
  81.                                  int coupled_stream_count,
  82.                                  const uint8_t *channel_mapping)
  83. {
  84.     uint8_t *p   = avctx->extradata;
  85.     int channels = avctx->channels;
  86.  
  87.     bytestream_put_buffer(&p, "OpusHead", 8);
  88.     bytestream_put_byte(&p, 1); /* Version */
  89.     bytestream_put_byte(&p, channels);
  90.     bytestream_put_le16(&p, avctx->delay); /* Lookahead samples at 48kHz */
  91.     bytestream_put_le32(&p, avctx->sample_rate); /* Original sample rate */
  92.     bytestream_put_le16(&p, 0); /* Gain of 0dB is recommended. */
  93.  
  94.     /* Channel mapping */
  95.     if (channels > 2) {
  96.         bytestream_put_byte(&p, channels <= 8 ? 1 : 255);
  97.         bytestream_put_byte(&p, stream_count);
  98.         bytestream_put_byte(&p, coupled_stream_count);
  99.         bytestream_put_buffer(&p, channel_mapping, channels);
  100.     } else {
  101.         bytestream_put_byte(&p, 0);
  102.     }
  103. }
  104.  
  105. static int libopus_configure_encoder(AVCodecContext *avctx, OpusMSEncoder *enc,
  106.                                      LibopusEncOpts *opts)
  107. {
  108.     int ret;
  109.  
  110.     if (avctx->global_quality) {
  111.         av_log(avctx, AV_LOG_ERROR,
  112.                "Quality-based encoding not supported, "
  113.                "please specify a bitrate and VBR setting.\n");
  114.         return AVERROR(EINVAL);
  115.     }
  116.  
  117.     ret = opus_multistream_encoder_ctl(enc, OPUS_SET_BITRATE(avctx->bit_rate));
  118.     if (ret != OPUS_OK) {
  119.         av_log(avctx, AV_LOG_ERROR,
  120.                "Failed to set bitrate: %s\n", opus_strerror(ret));
  121.         return ret;
  122.     }
  123.  
  124.     ret = opus_multistream_encoder_ctl(enc,
  125.                                        OPUS_SET_COMPLEXITY(opts->complexity));
  126.     if (ret != OPUS_OK)
  127.         av_log(avctx, AV_LOG_WARNING,
  128.                "Unable to set complexity: %s\n", opus_strerror(ret));
  129.  
  130.     ret = opus_multistream_encoder_ctl(enc, OPUS_SET_VBR(!!opts->vbr));
  131.     if (ret != OPUS_OK)
  132.         av_log(avctx, AV_LOG_WARNING,
  133.                "Unable to set VBR: %s\n", opus_strerror(ret));
  134.  
  135.     ret = opus_multistream_encoder_ctl(enc,
  136.                                        OPUS_SET_VBR_CONSTRAINT(opts->vbr == 2));
  137.     if (ret != OPUS_OK)
  138.         av_log(avctx, AV_LOG_WARNING,
  139.                "Unable to set constrained VBR: %s\n", opus_strerror(ret));
  140.  
  141.     ret = opus_multistream_encoder_ctl(enc,
  142.                                        OPUS_SET_PACKET_LOSS_PERC(opts->packet_loss));
  143.     if (ret != OPUS_OK)
  144.         av_log(avctx, AV_LOG_WARNING,
  145.                "Unable to set expected packet loss percentage: %s\n",
  146.                opus_strerror(ret));
  147.  
  148.     if (avctx->cutoff) {
  149.         ret = opus_multistream_encoder_ctl(enc,
  150.                                            OPUS_SET_MAX_BANDWIDTH(opts->max_bandwidth));
  151.         if (ret != OPUS_OK)
  152.             av_log(avctx, AV_LOG_WARNING,
  153.                    "Unable to set maximum bandwidth: %s\n", opus_strerror(ret));
  154.     }
  155.  
  156.     return OPUS_OK;
  157. }
  158.  
  159. static av_cold int libopus_encode_init(AVCodecContext *avctx)
  160. {
  161.     LibopusEncContext *opus = avctx->priv_data;
  162.     const uint8_t *channel_mapping;
  163.     OpusMSEncoder *enc;
  164.     int ret = OPUS_OK;
  165.     int coupled_stream_count, header_size, frame_size;
  166.  
  167.     coupled_stream_count = opus_coupled_streams[avctx->channels - 1];
  168.     opus->stream_count   = avctx->channels - coupled_stream_count;
  169.     channel_mapping      = libavcodec_libopus_channel_map[avctx->channels - 1];
  170.  
  171.     /* FIXME: Opus can handle up to 255 channels. However, the mapping for
  172.      * anything greater than 8 is undefined. */
  173.     if (avctx->channels > 8)
  174.         av_log(avctx, AV_LOG_WARNING,
  175.                "Channel layout undefined for %d channels.\n", avctx->channels);
  176.  
  177.     if (!avctx->bit_rate) {
  178.         /* Sane default copied from opusenc */
  179.         avctx->bit_rate = 64000 * opus->stream_count +
  180.                           32000 * coupled_stream_count;
  181.         av_log(avctx, AV_LOG_WARNING,
  182.                "No bit rate set. Defaulting to %d bps.\n", avctx->bit_rate);
  183.     }
  184.  
  185.     if (avctx->bit_rate < 500 || avctx->bit_rate > 256000 * avctx->channels) {
  186.         av_log(avctx, AV_LOG_ERROR, "The bit rate %d bps is unsupported. "
  187.                "Please choose a value between 500 and %d.\n", avctx->bit_rate,
  188.                256000 * avctx->channels);
  189.         return AVERROR(EINVAL);
  190.     }
  191.  
  192.     frame_size = opus->opts.frame_duration * 48000 / 1000;
  193.     switch (frame_size) {
  194.     case 120:
  195.     case 240:
  196.         if (opus->opts.application != OPUS_APPLICATION_RESTRICTED_LOWDELAY)
  197.             av_log(avctx, AV_LOG_WARNING,
  198.                    "LPC mode cannot be used with a frame duration of less "
  199.                    "than 10ms. Enabling restricted low-delay mode.\n"
  200.                    "Use a longer frame duration if this is not what you want.\n");
  201.         /* Frame sizes less than 10 ms can only use MDCT mode, so switching to
  202.          * RESTRICTED_LOWDELAY avoids an unnecessary extra 2.5ms lookahead. */
  203.         opus->opts.application = OPUS_APPLICATION_RESTRICTED_LOWDELAY;
  204.     case 480:
  205.     case 960:
  206.     case 1920:
  207.     case 2880:
  208.         opus->opts.packet_size =
  209.         avctx->frame_size      = frame_size * avctx->sample_rate / 48000;
  210.         break;
  211.     default:
  212.         av_log(avctx, AV_LOG_ERROR, "Invalid frame duration: %g.\n"
  213.                "Frame duration must be exactly one of: 2.5, 5, 10, 20, 40 or 60.\n",
  214.                opus->opts.frame_duration);
  215.         return AVERROR(EINVAL);
  216.     }
  217.  
  218.     if (avctx->compression_level < 0 || avctx->compression_level > 10) {
  219.         av_log(avctx, AV_LOG_WARNING,
  220.                "Compression level must be in the range 0 to 10. "
  221.                "Defaulting to 10.\n");
  222.         opus->opts.complexity = 10;
  223.     } else {
  224.         opus->opts.complexity = avctx->compression_level;
  225.     }
  226.  
  227.     if (avctx->cutoff) {
  228.         switch (avctx->cutoff) {
  229.         case  4000:
  230.             opus->opts.max_bandwidth = OPUS_BANDWIDTH_NARROWBAND;
  231.             break;
  232.         case  6000:
  233.             opus->opts.max_bandwidth = OPUS_BANDWIDTH_MEDIUMBAND;
  234.             break;
  235.         case  8000:
  236.             opus->opts.max_bandwidth = OPUS_BANDWIDTH_WIDEBAND;
  237.             break;
  238.         case 12000:
  239.             opus->opts.max_bandwidth = OPUS_BANDWIDTH_SUPERWIDEBAND;
  240.             break;
  241.         case 20000:
  242.             opus->opts.max_bandwidth = OPUS_BANDWIDTH_FULLBAND;
  243.             break;
  244.         default:
  245.             av_log(avctx, AV_LOG_WARNING,
  246.                    "Invalid frequency cutoff: %d. Using default maximum bandwidth.\n"
  247.                    "Cutoff frequency must be exactly one of: 4000, 6000, 8000, 12000 or 20000.\n",
  248.                    avctx->cutoff);
  249.             avctx->cutoff = 0;
  250.         }
  251.     }
  252.  
  253.     enc = opus_multistream_encoder_create(avctx->sample_rate, avctx->channels,
  254.                                           opus->stream_count,
  255.                                           coupled_stream_count,
  256.                                           channel_mapping,
  257.                                           opus->opts.application, &ret);
  258.     if (ret != OPUS_OK) {
  259.         av_log(avctx, AV_LOG_ERROR,
  260.                "Failed to create encoder: %s\n", opus_strerror(ret));
  261.         return ff_opus_error_to_averror(ret);
  262.     }
  263.  
  264.     ret = libopus_configure_encoder(avctx, enc, &opus->opts);
  265.     if (ret != OPUS_OK) {
  266.         ret = ff_opus_error_to_averror(ret);
  267.         goto fail;
  268.     }
  269.  
  270.     header_size = 19 + (avctx->channels > 2 ? 2 + avctx->channels : 0);
  271.     avctx->extradata = av_malloc(header_size + FF_INPUT_BUFFER_PADDING_SIZE);
  272.     if (!avctx->extradata) {
  273.         av_log(avctx, AV_LOG_ERROR, "Failed to allocate extradata.\n");
  274.         ret = AVERROR(ENOMEM);
  275.         goto fail;
  276.     }
  277.     avctx->extradata_size = header_size;
  278.  
  279.     opus->samples = av_mallocz(frame_size * avctx->channels *
  280.                                av_get_bytes_per_sample(avctx->sample_fmt));
  281.     if (!opus->samples) {
  282.         av_log(avctx, AV_LOG_ERROR, "Failed to allocate samples buffer.\n");
  283.         ret = AVERROR(ENOMEM);
  284.         goto fail;
  285.     }
  286.  
  287.     ret = opus_multistream_encoder_ctl(enc, OPUS_GET_LOOKAHEAD(&avctx->delay));
  288.     if (ret != OPUS_OK)
  289.         av_log(avctx, AV_LOG_WARNING,
  290.                "Unable to get number of lookahead samples: %s\n",
  291.                opus_strerror(ret));
  292.  
  293.     libopus_write_header(avctx, opus->stream_count, coupled_stream_count,
  294.                          opus_vorbis_channel_map[avctx->channels - 1]);
  295.  
  296.     ff_af_queue_init(avctx, &opus->afq);
  297.  
  298.     opus->enc = enc;
  299.  
  300.     return 0;
  301.  
  302. fail:
  303.     opus_multistream_encoder_destroy(enc);
  304.     av_freep(&avctx->extradata);
  305.     return ret;
  306. }
  307.  
  308. static int libopus_encode(AVCodecContext *avctx, AVPacket *avpkt,
  309.                           const AVFrame *frame, int *got_packet_ptr)
  310. {
  311.     LibopusEncContext *opus = avctx->priv_data;
  312.     const int sample_size   = avctx->channels *
  313.                               av_get_bytes_per_sample(avctx->sample_fmt);
  314.     uint8_t *audio;
  315.     int ret;
  316.     int discard_padding;
  317.  
  318.     if (frame) {
  319.         ff_af_queue_add(&opus->afq, frame);
  320.         if (frame->nb_samples < opus->opts.packet_size) {
  321.             audio = opus->samples;
  322.             memcpy(audio, frame->data[0], frame->nb_samples * sample_size);
  323.         } else
  324.             audio = frame->data[0];
  325.     } else {
  326.         if (!opus->afq.remaining_samples)
  327.             return 0;
  328.         audio = opus->samples;
  329.         memset(audio, 0, opus->opts.packet_size * sample_size);
  330.     }
  331.  
  332.     /* Maximum packet size taken from opusenc in opus-tools. 60ms packets
  333.      * consist of 3 frames in one packet. The maximum frame size is 1275
  334.      * bytes along with the largest possible packet header of 7 bytes. */
  335.     if ((ret = ff_alloc_packet2(avctx, avpkt, (1275 * 3 + 7) * opus->stream_count)) < 0)
  336.         return ret;
  337.  
  338.     if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT)
  339.         ret = opus_multistream_encode_float(opus->enc, (float *)audio,
  340.                                             opus->opts.packet_size,
  341.                                             avpkt->data, avpkt->size);
  342.     else
  343.         ret = opus_multistream_encode(opus->enc, (opus_int16 *)audio,
  344.                                       opus->opts.packet_size,
  345.                                       avpkt->data, avpkt->size);
  346.  
  347.     if (ret < 0) {
  348.         av_log(avctx, AV_LOG_ERROR,
  349.                "Error encoding frame: %s\n", opus_strerror(ret));
  350.         return ff_opus_error_to_averror(ret);
  351.     }
  352.  
  353.     av_shrink_packet(avpkt, ret);
  354.  
  355.     ff_af_queue_remove(&opus->afq, opus->opts.packet_size,
  356.                        &avpkt->pts, &avpkt->duration);
  357.  
  358.     discard_padding = opus->opts.packet_size - avpkt->duration;
  359.     // Check if subtraction resulted in an overflow
  360.     if ((discard_padding < opus->opts.packet_size) != (avpkt->duration > 0)) {
  361.         av_free_packet(avpkt);
  362.         av_free(avpkt);
  363.         return AVERROR(EINVAL);
  364.     }
  365.     if (discard_padding > 0) {
  366.         uint8_t* side_data = av_packet_new_side_data(avpkt,
  367.                                                      AV_PKT_DATA_SKIP_SAMPLES,
  368.                                                      10);
  369.         if(side_data == NULL) {
  370.             av_free_packet(avpkt);
  371.             av_free(avpkt);
  372.             return AVERROR(ENOMEM);
  373.         }
  374.         AV_WL32(side_data + 4, discard_padding);
  375.     }
  376.  
  377.     *got_packet_ptr = 1;
  378.  
  379.     return 0;
  380. }
  381.  
  382. static av_cold int libopus_encode_close(AVCodecContext *avctx)
  383. {
  384.     LibopusEncContext *opus = avctx->priv_data;
  385.  
  386.     opus_multistream_encoder_destroy(opus->enc);
  387.  
  388.     ff_af_queue_close(&opus->afq);
  389.  
  390.     av_freep(&opus->samples);
  391.     av_freep(&avctx->extradata);
  392.  
  393.     return 0;
  394. }
  395.  
  396. #define OFFSET(x) offsetof(LibopusEncContext, opts.x)
  397. #define FLAGS AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM
  398. static const AVOption libopus_options[] = {
  399.     { "application",    "Intended application type",           OFFSET(application),    AV_OPT_TYPE_INT,   { .i64 = OPUS_APPLICATION_AUDIO }, OPUS_APPLICATION_VOIP, OPUS_APPLICATION_RESTRICTED_LOWDELAY, FLAGS, "application" },
  400.         { "voip",           "Favor improved speech intelligibility",   0, AV_OPT_TYPE_CONST, { .i64 = OPUS_APPLICATION_VOIP },                0, 0, FLAGS, "application" },
  401.         { "audio",          "Favor faithfulness to the input",         0, AV_OPT_TYPE_CONST, { .i64 = OPUS_APPLICATION_AUDIO },               0, 0, FLAGS, "application" },
  402.         { "lowdelay",       "Restrict to only the lowest delay modes", 0, AV_OPT_TYPE_CONST, { .i64 = OPUS_APPLICATION_RESTRICTED_LOWDELAY }, 0, 0, FLAGS, "application" },
  403.     { "frame_duration", "Duration of a frame in milliseconds", OFFSET(frame_duration), AV_OPT_TYPE_FLOAT, { .dbl = 10.0 }, 2.5, 60.0, FLAGS },
  404.     { "packet_loss",    "Expected packet loss percentage",     OFFSET(packet_loss),    AV_OPT_TYPE_INT,   { .i64 = 0 },    0,   100,  FLAGS },
  405.     { "vbr",            "Variable bit rate mode",              OFFSET(vbr),            AV_OPT_TYPE_INT,   { .i64 = 1 },    0,   2,    FLAGS, "vbr" },
  406.         { "off",            "Use constant bit rate", 0, AV_OPT_TYPE_CONST, { .i64 = 0 }, 0, 0, FLAGS, "vbr" },
  407.         { "on",             "Use variable bit rate", 0, AV_OPT_TYPE_CONST, { .i64 = 1 }, 0, 0, FLAGS, "vbr" },
  408.         { "constrained",    "Use constrained VBR",   0, AV_OPT_TYPE_CONST, { .i64 = 2 }, 0, 0, FLAGS, "vbr" },
  409.     { NULL },
  410. };
  411.  
  412. static const AVClass libopus_class = {
  413.     .class_name = "libopus",
  414.     .item_name  = av_default_item_name,
  415.     .option     = libopus_options,
  416.     .version    = LIBAVUTIL_VERSION_INT,
  417. };
  418.  
  419. static const AVCodecDefault libopus_defaults[] = {
  420.     { "b",                 "0" },
  421.     { "compression_level", "10" },
  422.     { NULL },
  423. };
  424.  
  425. static const int libopus_sample_rates[] = {
  426.     48000, 24000, 16000, 12000, 8000, 0,
  427. };
  428.  
  429. AVCodec ff_libopus_encoder = {
  430.     .name            = "libopus",
  431.     .long_name       = NULL_IF_CONFIG_SMALL("libopus Opus"),
  432.     .type            = AVMEDIA_TYPE_AUDIO,
  433.     .id              = AV_CODEC_ID_OPUS,
  434.     .priv_data_size  = sizeof(LibopusEncContext),
  435.     .init            = libopus_encode_init,
  436.     .encode2         = libopus_encode,
  437.     .close           = libopus_encode_close,
  438.     .capabilities    = CODEC_CAP_DELAY | CODEC_CAP_SMALL_LAST_FRAME,
  439.     .sample_fmts     = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
  440.                                                       AV_SAMPLE_FMT_FLT,
  441.                                                       AV_SAMPLE_FMT_NONE },
  442.     .channel_layouts = ff_vorbis_channel_layouts,
  443.     .supported_samplerates = libopus_sample_rates,
  444.     .priv_class      = &libopus_class,
  445.     .defaults        = libopus_defaults,
  446. };
  447.